[asterisk-commits] tilghman: tag 1.6.0.14-rc1 r211721 - /tags/1.6.0.14-rc1/

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Aug 11 16:26:08 CDT 2009


Author: tilghman
Date: Tue Aug 11 16:26:03 2009
New Revision: 211721

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=211721
Log:
Importing files for 1.6.0.14-rc1 release.

Added:
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    tags/1.6.0.14-rc1/.version   (with props)
    tags/1.6.0.14-rc1/ChangeLog   (with props)

Added: tags/1.6.0.14-rc1/.lastclean
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Added: tags/1.6.0.14-rc1/ChangeLog
URL: http://svn.asterisk.org/svn-view/asterisk/tags/1.6.0.14-rc1/ChangeLog?view=auto&rev=211721
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--- tags/1.6.0.14-rc1/ChangeLog (added)
+++ tags/1.6.0.14-rc1/ChangeLog Tue Aug 11 16:26:03 2009
@@ -1,0 +1,52815 @@
+2009-08-11  Tilghman Lesher <tlesher at digium.com>
+
+	* Released 1.6.0.14-rc1
+
+2009-08-10 19:51 +0000 [r211551-211587]  Tilghman Lesher <tlesher at digium.com>
+
+	* doc/CODING-GUIDELINES, /: Merged revisions 211584 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r211584 | tilghman | 2009-08-10 14:49:41 -0500
+	  (Mon, 10 Aug 2009) | 9 lines Merged revisions 211583 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/branches/1.4
+	  ........ r211583 | tilghman | 2009-08-10 14:48:48 -0500 (Mon, 10
+	  Aug 2009) | 1 line Conversion specifiers, not format specifiers
+	  ........ ................
+
+	* res/res_config_curl.c, apps/app_waitforring.c,
+	  channels/chan_misdn.c, funcs/func_channel.c, apps/app_macro.c,
+	  pbx/pbx_config.c, apps/app_chanspy.c, apps/app_mixmonitor.c,
+	  main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c,
+	  doc/CODING-GUIDELINES, utils/muted.c, apps/app_meetme.c,
+	  main/utils.c, cdr/cdr_pgsql.c, res/res_musiconhold.c,
+	  apps/app_followme.c, channels/misdn_config.c, utils/frame.c,
+	  res/ael/pval.c, main/cdr.c, main/channel.c, funcs/func_enum.c,
+	  channels/chan_phone.c, apps/app_setcallerid.c,
+	  apps/app_osplookup.c, main/manager.c, funcs/func_odbc.c,
+	  res/res_agi.c, apps/app_minivm.c, res/res_config_ldap.c,
+	  apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c,
+	  funcs/func_dialplan.c, main/dnsmgr.c, channels/chan_sip.c,
+	  res/res_limit.c, apps/app_waitforsilence.c, agi/eagi-test.c,
+	  main/acl.c, apps/app_waituntil.c, apps/app_queue.c,
+	  channels/chan_oss.c, agi/eagi-sphinx-test.c,
+	  channels/chan_usbradio.c, res/snmp/agent.c, pbx/pbx_dundi.c,
+	  apps/app_sms.c, utils/extconf.c, apps/app_verbose.c,
+	  apps/app_stack.c, apps/app_dahdibarge.c, funcs/func_rand.c,
+	  main/frame.c, apps/app_readfile.c, /, apps/app_record.c,
+	  funcs/func_strings.c, apps/app_alarmreceiver.c,
+	  cdr/cdr_adaptive_odbc.c, channels/chan_iax2.c,
+	  main/indications.c, main/config.c, main/cli.c,
+	  pbx/pbx_loopback.c, channels/chan_dahdi.c, pbx/pbx_spool.c,
+	  res/res_smdi.c, channels/chan_skinny.c, main/features.c,
+	  main/http.c, main/pbx.c, apps/app_privacy.c,
+	  codecs/codec_speex.c, channels/chan_agent.c, funcs/func_math.c,
+	  apps/app_disa.c, apps/app_morsecode.c, channels/iax2-provision.c,
+	  funcs/func_cut.c, pbx/dundi-parser.c, apps/app_talkdetect.c:
+	  AST-2009-005
+
+2009-08-10 14:10 +0000 [r211348]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 211347 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r211347 |
+	  file | 2009-08-10 11:07:44 -0300 (Mon, 10 Aug 2009) | 5 lines Fix
+	  retrieval of the port used for the video stream when adding SDP
+	  to a SIP message. (closes issue #15121) Reported by: jsmith
+	  ........
+
+2009-08-09 15:43 +0000 [r211233-211276]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, main/astfd.c: Merged revisions 211275 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r211275 | tilghman | 2009-08-09 10:42:02 -0500 (Sun, 09 Aug 2009)
+	  | 9 lines Merged revisions 211274 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r211274 | tilghman | 2009-08-09 10:41:01 -0500 (Sun, 09 Aug 2009)
+	  | 2 lines Small oops. Clear the flags which have been checked.
+	  ........ ................
+
+	* apps/app_stack.c, /: Merged revisions 211232 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r211232 |
+	  tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
+	  Check for NULL frame, before dereferencing pointer. (closes issue
+	  #15617) Reported by: rain ........
+
+2009-08-07 20:14 +0000 [r211114]  Russell Bryant <russell at digium.com>
+
+	* apps/app_chanspy.c, /: Merged revisions 211113 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009)
+	  | 11 lines Recorded merge of revisions 211112 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009)
+	  | 4 lines Resolve a deadlock involving app_chanspy and
+	  masquerades. (ABE-1936) ........ ................
+
+2009-08-07 18:18 +0000 [r211044]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, apps/app_queue.c: Merged revisions 211040 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009)
+	  | 21 lines Merged revisions 211038 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009)
+	  | 14 lines QUEUE_MEMBER_LIST _really_ wants the interface name,
+	  not the membername. This is a partial revert of revision 82590,
+	  which was an attempted cleanup, but in reality, it broke
+	  QUEUE_MEMBER_LIST, which has always been intended as a method by
+	  which component interfaces could be queried from the queue.
+	  Membername isn't useful here, because that field cannot be used
+	  to obtain further information about the member. See the
+	  documentation on QUEUE_MEMBER_LIST, RemoveQueueMember,
+	  QUEUE_MEMBER_PENALTY, and the various AMI commands which take a
+	  member argument for further justification. (closes issue #15664)
+	  Reported by: rain Patches: app_queue-queue_member_list.diff
+	  uploaded by rain (license 327) ........ ................
+
+2009-08-07 13:08 +0000 [r210993]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/udptl.c, /: Merged revisions 210992 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r210992 |
+	  kpfleming | 2009-08-07 08:08:00 -0500 (Fri, 07 Aug 2009) | 13
+	  lines Workaround broken T.38 endpoints that offer tiny
+	  MaxDatagram sizes. Some T.38 endpoints treat T38FaxMaxDatagram as
+	  the maximum IFP size that should be sent to them, rather than the
+	  maximum packet payload size. If such an endpoint also requests
+	  UDPRedundancy as the error correction mode, we'll end up
+	  calculating a tiny maximum IFP size, so small as to be unusable.
+	  This patch sets a lower bound on what we'll consider the remote's
+	  maximum IFP size to be, assuming that endpoints that do this
+	  really can accept larger packets than they've offered to accept.
+	  (closes issue #15649) Reported by: dazza76 ........
+
+2009-08-06 21:46 +0000 [r210909-210915]  Tilghman Lesher <tlesher at digium.com>
+
+	* main/channel.c, /: Merged revisions 210914 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r210914 | tilghman | 2009-08-06 16:46:01 -0500 (Thu, 06 Aug 2009)
+	  | 14 lines Merged revisions 210913 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210913 | tilghman | 2009-08-06 16:45:01 -0500 (Thu, 06 Aug 2009)
+	  | 7 lines Because channel information can be accessed outside of
+	  the channel thread, we must lock the channel prior to modifying
+	  it. (closes issue #15397) Reported by: caspy Patches:
+	  20090714__issue15397.diff.txt uploaded by tilghman (license 14)
+	  Tested by: caspy ........ ................
+
+	* apps/app_stack.c, include/asterisk/app.h, /, main/app.c: Merged
+	  revisions 210908 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r210908 |
+	  tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
+	  Allow Gosub to recognize quote delimiters without consuming them.
+	  (closes issue #15557) Reported by: rain Patches:
+	  20090723__issue15557.diff.txt uploaded by tilghman (license 14)
+	  Tested by: rain Review: https://reviewboard.asterisk.org/r/316/
+	  ........
+
+2009-08-06 17:47 +0000 [r210818]  Joshua Colp <jcolp at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 210817 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r210817 |
+	  file | 2009-08-06 14:47:04 -0300 (Thu, 06 Aug 2009) | 11 lines
+	  Accept additional T.38 reinvites after an initial one has been
+	  handled. Discussion of this subject has yielded that it is not
+	  actually acceptable to change T.38 parameters after the initial
+	  reinvite but declining is harsh and can cause the fax to fail
+	  when it may be possible to allow it to continue. This patch
+	  changes things so that additional T.38 reinvites are accepted but
+	  parameter changes ignored. This gives the fax a fighting chance.
+	  (closes issue #15610) Reported by: huangtx2009 ........
+
+2009-08-05 20:07 +0000 [r210647]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 210640 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r210640 | rmudgett | 2009-08-05 14:40:03 -0500
+	  (Wed, 05 Aug 2009) | 21 lines Merged revisions 210575 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210575 | rmudgett | 2009-08-05 14:18:56 -0500 (Wed, 05 Aug 2009)
+	  | 14 lines Dialplan starts execution before the channel setup is
+	  complete. * Issue 15655: For the case where dialing is complete
+	  for an incoming call, dahdi_new() was asked to start the PBX and
+	  then the code set more channel variables. If the dialplan hungup
+	  before these channel variables got set, asterisk would likely
+	  crash. * Fixed potential for overlap incoming call to erroneously
+	  set channel variables as global dialplan variables if the
+	  ast_channel structure failed to get allocated. * Added missing
+	  set of CALLINGSUBADDR in the dialing is complete case. (closes
+	  issue #15655) Reported by: alecdavis ........ ................
+
+2009-08-05 18:57 +0000 [r210568]  Leif Madsen <lmadsen at digium.com>
+
+	* doc/tex/imapstorage.tex, /: Merged revisions 210564 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r210564 | lmadsen | 2009-08-05 13:49:58 -0500
+	  (Wed, 05 Aug 2009) | 19 lines Merged revisions 210563 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210563 | lmadsen | 2009-08-05 13:46:21 -0500 (Wed, 05 Aug 2009)
+	  | 11 lines Update imapstorage.txt documentation. Updated the
+	  imapstorage.txt documentation to reflect that issues with
+	  c-client versions older than 2007 seem to cause crashing issues
+	  that are not seen with more recent versions. Documentation has
+	  been updated to reflect this. (closes issue #14496) Reported by:
+	  vbcrlfuser Patches: __20090727-imap-documentation-patch.txt
+	  uploaded by lmadsen (license 10) Tested by: lmadsen, mmichelson,
+	  dbrooks ........ ................
+
+2009-08-04 14:54 +0000 [r210239]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile, /: Merged revisions 210238 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r210238 | kpfleming | 2009-08-04 09:53:00 -0500 (Tue, 04 Aug
+	  2009) | 16 lines Merged revisions 210237 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r210237 | kpfleming | 2009-08-04 09:51:39 -0500 (Tue, 04 Aug
+	  2009) | 10 lines Eliminate spurious compiler warnings from system
+	  headers on *BSD platforms. Ensure that system headers located in
+	  /usr/local/include are actually treated as system headers by the
+	  compiler, and not as local headers which are subject to warnings
+	  from the -Wundef compiler option and others. (closes issue
+	  #15606) Reported by: mvanbaak ........ ................
+
+2009-08-01 11:31 +0000 [r209840-209896]  Russell Bryant <russell at digium.com>
+
+	* main/db1-ast/mpool/mpool.c, /: Merged revisions 209887 via
+	  svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r209887 | russell | 2009-08-01 06:29:25 -0500
+	  (Sat, 01 Aug 2009) | 12 lines Merged revisions 209879 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209879 | russell | 2009-08-01 06:27:25 -0500 (Sat, 01 Aug 2009)
+	  | 5 lines Resolve a valgrind warning about a read from
+	  uninitialized memory. (issue #15396) Reported by: aragon ........
+	  ................
+
+	* apps/app_milliwatt.c, /: Merged revisions 209839 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r209839 | russell | 2009-08-01 06:02:07 -0500
+	  (Sat, 01 Aug 2009) | 20 lines Merged revisions 209838 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009)
+	  | 13 lines Modify how Playtones() is used in Milliwatt() to
+	  resolve gain issue. When Milliwatt() was changed internally to
+	  use Playtones() so that the proper tone was used, it introduced a
+	  drop in gain in the output signal. So, use the playtones API
+	  directly and specify a volume argument such that the output
+	  matches the gain of the original Milliwatt() code. (closes issue
+	  #15386) Reported by: rue_mohr Patches: issue_15386.rev2.diff
+	  uploaded by russell (license 2) Tested by: rue_mohr ........
+	  ................
+
+2009-08-01 01:13 +0000 [r209762]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* channels/misdn/isdn_lib.c, utils/frame.c, /, main/Makefile,
+	  channels/misdn/ie.c, main/event.c: Merged revisions 209760-209761
+	  via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r209760 | kpfleming | 2009-07-31 20:03:07 -0500
+	  (Fri, 31 Jul 2009) | 13 lines Merged revisions 209759 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209759 | kpfleming | 2009-07-31 19:52:00 -0500 (Fri, 31 Jul
+	  2009) | 7 lines Minor changes inspired by testing with latest
+	  GCC. The latest GCC (what will become 4.5.x) has a few new
+	  warnings, that in these cases found some either downright buggy
+	  code, or at least seriously poorly designed code that could be
+	  improved. ........ ................ r209761 | kpfleming |
+	  2009-07-31 20:04:06 -0500 (Fri, 31 Jul 2009) | 1 line Revert
+	  accidental Makefile change. ................
+
+2009-07-31 21:56 +0000 [r209712]  Russell Bryant <russell at digium.com>
+
+	* /, main/event.c: Merged revisions 209711 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209711 |
+	  russell | 2009-07-31 16:53:31 -0500 (Fri, 31 Jul 2009) | 2 lines
+	  Fix some places where ast_event_type was used instead of
+	  ast_event_ie_type. ........
+
+2009-07-30 16:37 +0000 [r209555-209587]  David Brooks <dbrooks at digium.com>
+
+	* channels/chan_dahdi.c: Merged revisions 209554 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+	  dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+	  Fixes numerous spelling errors. Patch submitted by alecdavis.
+	  (closes issue #15595) Reported by: alecdavis ........
+
+	* include/asterisk/abstract_jb.h,
+	  contrib/init.d/rc.debian.asterisk, /, apps/app_sms.c,
+	  codecs/lpc10/pitsyn.c, channels/chan_console.c: Merged revisions
+	  209554 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209554 |
+	  dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
+	  Fixes numerous spelling errors. Patch submitted by alecdavis.
+	  (closes issue #15595) Reported by: alecdavis ........
+
+2009-07-28 12:01 +0000 [r209394]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* apps/app_fax.c: Correct error in backport of latest app_fax
+	  fixes.
+
+2009-07-28 00:19 +0000 [r209325]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, sounds/sounds.xml: Merged revisions 209317 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r209317 | tilghman | 2009-07-27 19:14:12 -0500 (Mon, 27 Jul 2009)
+	  | 9 lines Merged revisions 209315 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209315 | tilghman | 2009-07-27 19:12:03 -0500 (Mon, 27 Jul 2009)
+	  | 2 lines Publish French extra sounds ........ ................
+
+2009-07-27 21:44 +0000 [r209259-209280]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* /, apps/app_fax.c: Merged revisions 209279 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209279 |
+	  kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7
+	  lines Cleanup T.38 negotiation changes. Convert LOG_NOTICE
+	  messages about T.38 negotiation in debug level 1 messages, clean
+	  up some looping logic, and correct an improper use of ast_free()
+	  for freeing an ast_frame. ........
+
+	* /, apps/app_fax.c: Merged revisions 209256 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209256 |
+	  kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10
+	  lines Make T.38 switchover in ReceiveFAX synchronous. In receive
+	  mode, if the channel that ReceiveFAX is running on supports T.38,
+	  we should *always* attempt to switch T.38, rather than listening
+	  for an incoming CNG tone and only triggering on that. The channel
+	  may be using a low-bitrate codec that distorts the CNG tone, the
+	  sending FAX endpoint may not send CNG at all, or there could be a
+	  variety of other reasons that we don't detect it, but in all
+	  those cases if T.38 is available we certainly want to use it.
+	  ........
+
+2009-07-27 20:23 +0000 [r209221]  David Brooks <dbrooks at digium.com>
+
+	* channels/chan_dahdi.c, channels/chan_vpb.cc, res/res_smdi.c, /,
+	  include/asterisk/module.h, main/features.c, res/res_agi.c,
+	  res/res_jabber.c: Merged revisions 209098 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209098 |
+	  dbrooks | 2009-07-27 11:33:50 -0500 (Mon, 27 Jul 2009) | 6 lines
+	  Fixing typos. Replaces "recieved" with "received" and "initilize"
+	  with "initialize" (closes issue #15571) Reported by: alecdavis
+	  ........
+
+2009-07-27 20:16 +0000 [r209133-209198]  Mark Michelson <mmichelson at digium.com>
+
+	* /, res/res_musiconhold.c: Merged revisions 209197 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk ........
+	  r209197 | mmichelson | 2009-07-27 15:11:42 -0500 (Mon, 27 Jul
+	  2009) | 9 lines Honor channel's music class when using realtime
+	  music on hold. (closes issue #15051) Reported by: alexh Patches:
+	  15051.patch uploaded by mmichelson (license 60) Tested by: alexh
+	  ........
+
+	* main/udptl.c, /, configs/udptl.conf.sample: Merged revisions
+	  209132 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r209132 | mmichelson | 2009-07-27 12:50:04 -0500 (Mon, 27 Jul
+	  2009) | 24 lines Merged revisions 209131 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r209131 | mmichelson | 2009-07-27 12:44:06 -0500 (Mon, 27 Jul
+	  2009) | 18 lines Allow for UDPTL to use only even-numbered ports
+	  if desired. There are some VoIP providers out there that will not
+	  accept SDP offers with odd numbered UDPTL ports. While it is my
+	  personal opinion that these VoIP providers are misinterpreting
+	  RFC 2327, it really is not a big deal to play along with their
+	  silly little games. Of course, since restricting UDPTL ports to
+	  only even numbers reduces the range of available ports by half,
+	  so the option to use only even port numbers is off by default. A
+	  user can enable the behavior by setting use_even_ports=yes in
+	  udptl.conf. (closes issue #15182) Reported by: CGMChris Patches:
+	  15182.patch uploaded by mmichelson (license 60) Tested by:
+	  CGMChris ........ ................
+
+2009-07-27 16:06 +0000 [r209061]  David Brooks <dbrooks at digium.com>
+
+	* res/res_smdi.c, pbx/pbx_dundi.c: Just replacing typos "recieved"
+	  with "received". From issue #15360, forgot to apply to trunk and
+	  other branches.
+
+2009-07-27 15:39 +0000 [r209057]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* Makefile, /: Merged revisions 209056 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r209056 |
+	  kpfleming | 2009-07-27 10:38:59 -0500 (Mon, 27 Jul 2009) | 10
+	  lines Restore explicit export of ASTCFLAGS/ASTLDFLAGS and
+	  underscore-variants to sub-makes. During the recent Makefile
+	  improvements I made, it seemed the 'make' was automatically
+	  carrying down the ASTCFLAGS/ASTLDFLAGS settings to sub-makes, so
+	  I removed the explict export of them. However, there are some
+	  circumstances where make does this, and some where it does not,
+	  so I've brought them back to ensure they are always exported. I
+	  also removed an extraneous double setting of _ASTLDFLAGS on *BSD
+	  platforms. ........
+
+2009-07-27 01:21 +0000 [r208925]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, main/translate.c, channels/chan_iax2.c: Merged revisions
+	  208924 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r208924 | jpeeler | 2009-07-26 20:20:37 -0500 (Sun, 26 Jul 2009)
+	  | 9 lines Merged revisions 208923 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208923 | jpeeler | 2009-07-26 20:18:31 -0500 (Sun, 26 Jul 2009)
+	  | 2 lines Fix logic errors from 208746 ........ ................
+
+2009-07-25 06:24 +0000 [r208752]  Jeff Peeler <jpeeler at digium.com>
+
+	* /, channels/chan_skinny.c, main/translate.c,
+	  channels/chan_iax2.c: Merged revisions 208749 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r208749 | jpeeler | 2009-07-25 01:23:18 -0500 (Sat, 25 Jul 2009)
+	  | 13 lines Merged revisions 208746 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208746 | jpeeler | 2009-07-25 01:19:50 -0500 (Sat, 25 Jul 2009)
+	  | 7 lines Fix compiling under dev-mode with gcc 4.4.0. Mostly
+	  trivial changes, but I did not know of any other way to fix the
+	  "dereferencing type-punned pointer will break strict-aliasing
+	  rules" error without creating a tmp variable in chan_skinny.
+	  ........ ................
+
+2009-07-24 18:49 +0000 [r208594]  Russell Bryant <russell at digium.com>
+
+	* apps/app_dial.c, /: Merged revisions 208593 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009)
+	  | 14 lines Merged revisions 208592 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009)
+	  | 7 lines Do not log an ERROR if autoservice_stop() returns -1.
+	  This does not indicate an error. A return of -1 just means that
+	  the channel has been hung up. (reported in #asterisk-dev)
+	  ........ ................
+
+2009-07-24 18:31 +0000 [r208589]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 208588 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r208588 | mmichelson | 2009-07-24 13:31:04 -0500 (Fri, 24 Jul
+	  2009) | 16 lines Merged revisions 208587 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208587 | mmichelson | 2009-07-24 13:26:50 -0500 (Fri, 24 Jul
+	  2009) | 10 lines Only send a BYE when hanging up a channel that
+	  is up. For cases where Asterisk sends an INVITE and receives a
+	  non 2XX final response, Asterisk would follow the INVITE
+	  transaction by immediately sending a BYE, which was unnecessary.
+	  (closes issue #14575) Reported by: chris-mac ........
+	  ................
+
+2009-07-24 15:04 +0000 [r208468-208549]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* main/udptl.c, /, channels/chan_sip.c, include/asterisk/udptl.h:
+	  Merged revisions 208548 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r208548 |
+	  kpfleming | 2009-07-24 10:02:53 -0500 (Fri, 24 Jul 2009) | 8
+	  lines Resolve a T.38 negotiation issue left over from the
+	  udptl-updates merge. The udptl-updates branch that was merged
+	  yesterday failed to properly send back T.38 SDP responses with
+	  the correct error correction mode, if the incoming SDP from the
+	  other end caused us to change error correction modes. This patch
+	  corrects that situation. ........
+
+	* UPGRADE.txt: Use correct formatting for T.38 change note in
+	  UPGRADE.txt
+
+	* main/rtp.c, main/channel.c, main/udptl.c, main/frame.c, /,
+	  channels/chan_sip.c, apps/app_fax.c, UPGRADE.txt,
+	  include/asterisk/udptl.h, include/asterisk/frame.h: Merged
+	  revisions 208464 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 |
+	  kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46
+	  lines Rework of T.38 negotiation and UDPTL API to address
+	  interoperability problems Over the past couple of months, a
+	  number of issues with Asterisk negotiating (and successfully
+	  completing) T.38 sessions with various endpoints have been found.
+	  This patch attempts to address many of them, primarily focused
+	  around ensuring that the endpoints' MaxDatagram size is honored,
+	  and in addition by ensuring that T.38 session parameter
+	  negotiation is performed correctly according to the ITU T.38
+	  Recommendation. The major changes here are: 1) T.38 applications
+	  in Asterisk (app_fax) only generate/receive IFP packets, they do
+	  not ever work with UDPTL packets. As a result of this, they
+	  cannot be allowed to generate packets that would overflow the
+	  other endpoints' MaxDatagram size after the UDPTL stack adds any
+	  error correction information. With this patch, the application is
+	  told the maximum *IFP* size it can generate, based on a
+	  calculation using the far end MaxDatagram size and the active
+	  error correction mode on the T.38 session. The same is true for
+	  sending *our* MaxDatagram size to the remote endpoint; it is
+	  computed from the value that the application says it can accept
+	  (for a single IFP packet) combined with the active error
+	  correction mode. 2) All treatment of T.38 session parameters as
+	  'capabilities' in chan_sip has been removed; these parameters are
+	  not at all like audio/video stream capabilities. There are strict
+	  rules to follow for computing an answer to a T.38 offer, and
+	  chan_sip now follows those rules, using the desired parameters
+	  from the application (or channel) that wants to accept the T.38
+	  negotiation. 3) chan_sip now stores and forwards
+	  ast_control_t38_parameters structures for tracking 'our' and
+	  'their' T.38 session parameters; this greatly simplifies
+	  negotiation, especially for pass-through calls. 4) Since T.38
+	  negotiation without specifying parameters or receiving the final
+	  negotiated parameters is not very worthwhile, the AST_CONTROL_T38
+	  control frame has been removed. A note has been added to
+	  UPGRADE.txt about this removal, since any out-of-tree
+	  applications that use it will no longer function properly until
+	  they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review:
+	  https://reviewboard.asterisk.org/r/310/ ........
+
+2009-07-23 19:35 +0000 [r208389]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 208388 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r208388 | mmichelson | 2009-07-23 14:34:49 -0500 (Thu, 23 Jul
+	  2009) | 24 lines Merged revisions 208386 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208386 | mmichelson | 2009-07-23 14:24:21 -0500 (Thu, 23 Jul
+	  2009) | 17 lines Fix a problem where a 491 response could be sent
+	  out of dialog. This generalizes the fix for issue 13849. The
+	  initial fix corrected the problem that Asterisk would reply with
+	  a 491 if a reinvite were received from an endpoint and we had not
+	  yet received an ACK from that endpoint for the initial INVITE it
+	  had sent us. This expansion also allows Asterisk to appropriately
+	  handle an INVITE with authorization credentials if Asterisk had
+	  not received an ACK from the previous transaction in which
+	  Asterisk had responded to an unauthorized INVITE with a 407.
+	  (closes issue #14239) Reported by: klaus3000 Patches: 14239.patch
+	  uploaded by mmichelson (license 60) Tested by: klaus3000 ........
+	  ................
+
+2009-07-23 19:23 +0000 [r208384]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 208383 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r208383 | jpeeler | 2009-07-23 14:21:50 -0500
+	  (Thu, 23 Jul 2009) | 12 lines Merged revisions 208380 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208380 | jpeeler | 2009-07-23 14:19:53 -0500 (Thu, 23 Jul 2009)
+	  | 6 lines Only set the priindication setting when not performing
+	  a reload (closes issue #14696) Reported by: fdecher ........
+	  ................
+
+2009-07-23 16:30 +0000 [r208264-208316]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 208314 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r208314 | mmichelson | 2009-07-23 11:29:37 -0500 (Thu, 23 Jul
+	  2009) | 9 lines Merged revisions 208312 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208312 | mmichelson | 2009-07-23 11:29:18 -0500 (Thu, 23 Jul
+	  2009) | 3 lines Remove inaccurate XXX comment. ........
+	  ................
+
+	* /, channels/chan_sip.c: Merged revisions 208263 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r208263 | mmichelson | 2009-07-23 10:46:34 -0500 (Thu, 23 Jul
+	  2009) | 15 lines Merged revisions 208262 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r208262 | mmichelson | 2009-07-23 10:43:07 -0500 (Thu, 23 Jul
+	  2009) | 8 lines Properly handle 183 responses which do not
+	  contain an SDP. (closes issue #15442) Reported by: ffloimair
+	  Patches: 15442.patch uploaded by mmichelson (license 60) Tested
+	  by: tkarl, ffloimair ........ ................
+
+2009-07-21 22:47 +0000 [r207947]  Tilghman Lesher <tlesher at digium.com>
+
+	* /, funcs/func_strings.c: Merged revisions 207946 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r207946 | tilghman | 2009-07-21 17:45:32 -0500
+	  (Tue, 21 Jul 2009) | 15 lines Merged revisions 207945 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009)
+	  | 8 lines Force an error if a blank is passed to QUOTE (because
+	  the documentation states the argument is not optional). This
+	  change makes URIENCODE and QUOTE behave similarly, since the
+	  documentation states that the argument is not optional, for both.
+	  (closes issue #15439) Reported by: pkempgen Patches:
+	  20090706__issue15439.diff.txt uploaded by tilghman (license 14)
+	  ........ ................
+
+2009-07-21 20:27 +0000 [r207783-207860]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c, /: Merged revisions 207854 via svnmerge
+	  from https://origsvn.digium.com/svn/asterisk/trunk
+	  ................ r207854 | jpeeler | 2009-07-21 15:26:02 -0500
+	  (Tue, 21 Jul 2009) | 16 lines Merged revisions 207827 via
+	  svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r207827 | jpeeler | 2009-07-21 15:16:55 -0500 (Tue, 21 Jul 2009)
+	  | 9 lines Wait for wink before dialing when using E&M wink
+	  signaling There was already code for other signaling types in
+	  dahdi_handle_event to handle dialing if a dial operation dial
+	  string was present. Simply add SIG_EMWINK to the list. (closes
+	  issue #14434) Reported by: araasch ........ ................
+
+	* channels/chan_dahdi.c: Revert r207636, this approach could
+	  potentially block for an unacceptable amount of time.
+
+2009-07-21 14:30 +0000 [r207725]  Mark Michelson <mmichelson at digium.com>
+
+	* main/manager.c, /: Merged revisions 207723 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r207723 | mmichelson | 2009-07-21 09:29:40 -0500 (Tue, 21 Jul
+	  2009) | 11 lines Merged revisions 207714 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r207714 | mmichelson | 2009-07-21 09:26:00 -0500 (Tue, 21 Jul
+	  2009) | 5 lines Document default timeout for AMI originations.
+	  AST-224 ........ ................
+
+2009-07-21 13:39 +0000 [r207683]  Kevin P. Fleming <kpfleming at digium.com>
+
+	* funcs/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile,
+	  Makefile, agi/Makefile, codecs/Makefile, utils/Makefile, /,
+	  main/Makefile, codecs/gsm/Makefile, Makefile.moddir_rules,
+	  Makefile.rules, pbx/Makefile, res/Makefile, channels/Makefile:
+	  Merged revisions 207680 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r207680 | kpfleming | 2009-07-21 08:28:04 -0500 (Tue, 21 Jul
+	  2009) | 18 lines Merged revisions 207647 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul
+	  2009) | 12 lines Ensure that user-provided CFLAGS and LDFLAGS are
+	  honored. This commit changes the build system so that
+	  user-provided flags (in ASTCFLAGS and ASTLDFLAGS) are supplied to
+	  the compiler/linker *after* all flags provided by the build
+	  system itself, so that the user can effectively override the
+	  build system's flags if desired. In addition, ASTCFLAGS and
+	  ASTLDFLAGS can now be provided *either* in the environment before
+	  running 'make', or as variable assignments on the 'make' command
+	  line. As a result, the use of COPTS and LDOPTS is no longer
+	  necessary, so they are no longer documented, but are still
+	  supported so as not to break existing build systems that supply
+	  them when building Asterisk. ........ ................
+
+2009-07-21 04:38 +0000 [r207636]  Jeff Peeler <jpeeler at digium.com>
+
+	* channels/chan_dahdi.c: Wait for wink before dialing when using
+	  E&M wink signaling This patch adds a new dahdi_wait function to
+	  specifically wait for the wink event. If the wink is not
+	  eventually received the channel is hung up. (closes issue #14434)
+	  Reported by: araasch Patches: emwinkmod uploaded by araasch
+	  (license 693)
+
+2009-07-20 19:55 +0000 [r207425]  Mark Michelson <mmichelson at digium.com>
+
+	* /, channels/chan_sip.c: Merged revisions 207424 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r207424 | mmichelson | 2009-07-20 14:48:12 -0500 (Mon, 20 Jul
+	  2009) | 39 lines Merged revisions 207423 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r207423 | mmichelson | 2009-07-20 14:39:59 -0500 (Mon, 20 Jul
+	  2009) | 33 lines Answer video SDP offers properly when
+	  videosupport is not enabled. Copied from Review board: In issue
+	  12434, the reporter describes a situation in which audio and
+	  video is offered on the call, but because videosupport is
+	  disabled in sip.conf, Asterisk gives no response at all to the
+	  video offer. According to RFC 3264, all media offers should have
+	  a corresponding answer. For offers we do not intend to actually
+	  reply to with meaningful values, we should still reply with the
+	  port for the media stream set to 0. In this patch, we take note
+	  of what types of media have been offered and save the information
+	  on the sip_pvt. The SDP in the response will take into account
+	  whether media was offered. If we are not otherwise going to
+	  answer a media offer, we will insert an appropriate m= line with
+	  the port set to 0. It is important to note that this patch is
+	  pretty much a bandage being applied to a broken bone. The patch
+	  *only* helps for situations where video is offered but
+	  videosupport is disabled and when udptl_pt is disabled but T.38
+	  is offered. Asterisk is not guaranteed to respond to every media
+	  offer. Notable cases are when multiple streams of the same type
+	  are offered. The 2 media stream limit is still present with this
+	  patch, too. In trunk and the 1.6.X branches, things will be a bit
+	  different since Asterisk also supports text in SDPs as well.
+	  (closes issue #12434) Reported by: mnnojd Review:
+	  https://reviewboard.asterisk.org/r/311 Review:
+	  https://reviewboard.asterisk.org/r/313 ........ ................
+
+2009-07-20 16:37 +0000 [r207362]  Russell Bryant <russell at digium.com>
+
+	* main/channel.c, /: Merged revisions 207361 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/trunk ................
+	  r207361 | russell | 2009-07-20 11:36:15 -0500 (Mon, 20 Jul 2009)
+	  | 16 lines Merged revisions 207360 via svnmerge from
+	  https://origsvn.digium.com/svn/asterisk/branches/1.4 ........
+	  r207360 | russell | 2009-07-20 11:26:24 -0500 (Mon, 20 Jul 2009)
+	  | 9 lines Only do the chan->fdno check in ast_read() in a
+	  developer build. I changed this check to only happen in a
+	  dev-mode build. I also added a comment explaining what is going
+	  on. I also made it so that detection of this situation does not
+	  affect ast_read() operation. (closes issue #14723) Reported by:
+	  seadweller ........ ................
+
+2009-07-18 01:35 +0000 [r207286]  Richard Mudgett <rmudgett at digium.com>
+
+	* channels/misdn/isdn_lib.c, channels/misdn_config.c,
+	  channels/misdn/isdn_lib_intern.h, channels/misdn/isdn_lib.h,

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